This DSP component takes impulse response and does a fast convolution of the sound data with that impulse response.

Basically, it allows quick computation of any combination of linear effects, such as equalization, echo, flange, reverb, phase shift and so on. It can transform the sound to exactly the image you would get as where the impulse response wasrecorded, allowing you to get an accurate reproduction of a big theather or stadium.

This effect is available as 'Acoustic Mirror' in Sound Forge.A restricted version is available as 'Digital Convolution'in Cool Edit.

Question - what tools would you recommend for creating custom impulse responses, say for a room correction amplitude equalizer? Could this be done with Sound Forge?

I have no experience with 'room correction amplitude equalization', but basically, if you find a single tool that can do it, and allows you to feed the impulse through it, you're set.

You can even use analogue equipment.

Thanks, I think I can make this work for equalization. By the way, any chance this plugin could be made to have the option for separate impulse responses for each channel? I'm interested in equalization and that would help a lot.

Also, is it correct to assume that the impulse response must be created with the same sampling rate as the file to be convolved? If not I would expect the time/frequency values would shift.

Thanks, I think I can make this work for equalization. By the way, any chance this plugin could be made to have the option for separate impulse responses for each channel? I'm interested in equalization and that would help a lot.

Way ahead of you Just make stereo impulse responses.

QUOTE

Also, is it correct to assume that the impulse response must be created with the same sampling rate as the file to be convolved? If not I would expect the time/frequency values would shift.

As explained in the readme, just feed an impulse through your effect of choice. In this case, diskwrite 'Unitimpulse2k.wav' with EQ settings of choice enabled.

Ah, so it can't be done algorithmically?

How is this not 'algorithmically'???

You need to transform the frequency response of the EQ into an impulse response. Guess what diskwriting the impulse with EQ enabled does...

Edit: I guess that you're asking whether it's possible to make an EQ component that uses the foo_convolve engine as a backend for the equalizer. Yes. In fact I checked and there are similarities between the current backends, but my code is faster, and you can finetune the filter length. Not to mention it also works in stereo.

You need to transform the frequency response of the EQ into an impulse response. Guess what diskwriting the impulse with EQ enabled does...

Edit: I guess that you're asking whether it's possible to make an EQ component that uses the foo_convolve engine as a backend for the equalizer. Yes. In fact I checked and there are similarities between the current backends, but my code is faster, and you can finetune the filter length. Not to mention it also works in stereo.

I mean directly calculating the tap data, rather than running a Dirac pulse through a system and finding its impulse response.

Ack. I think I need to start being more specific when I'm posting about technical stuff...

(I took the DBX 160 file) open in wave editor, keep only first 16384 samples, duplicate channel to get stereo, then save & set as impulse file in FB2k. Play something, then close FB2k. Open it and play the thing again, but now go into DSP properties to the convolver settings. You'll be granted with either a crash or a very loud distorted sound (kinda like microphone feedback).

you don't think this type of thing would/should be an all or nothing affair? Seems to me that your impulse response would be giving you exactly what you wanted and there wouldn't be any need for mixing it with the original.

you don't think this type of thing would/should be an all or nothing affair? Seems to me that your impulse response would be giving you exactly what you wanted and there wouldn't be any need for mixing it with the original.

Well, yes, you're probably right.

I'm just used to wet/dry sliders in DSP effects Especially in reverb, hrtf-related effects and similiar stuff. It's just an easy and fast way to adjust "amount" of effect you'd like at the moment. It's not "a must" for convolver, but that was first thing that I've been kinda "missing" while I was playing with IRs converted from SF Acoustic Mirror

Guess it could also be general request for DSP manager, but there are some plugins (like gap killers, crosfaders...) that couldn't utilize this functionality for any reasonable purpose... So, no.